Puppy Phone - VOIP using SIP

Under development: PCMCIA, wireless, etc.
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smokey01
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#21 Post by smokey01 »

gcmartin wrote:Could someone share a location of the steps to connect one user with another. Is this just audio or is it video as well.? Just audio
Can a single SIP user connect to a SIP telephone on the LAN? Yes
Can a SIP user connect to a SIP device which can ring telephones in the home? Yes but you need to connect through a sip server that has a gateway to the PSTN telephone network. This will cost a small fee.
Can the SIP user connect to another SIP user via VPN? Not sure

Thanks in advance

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#22 Post by smokey01 »

Flash wrote:I don't understand why a router or NAT server doesn't seem to have the problems when I connect PSIP or another VOIP client to a server that it apparently has if I try to connect PSIP directly to another computer (with the NAT server or router between them.) If having a server between the two computers will make the NAT server or router happy, why not configure one of the servers that comes with Puppy to do the job, and then connect PSIP or whatever through it? I'm asking this in order to demonstrate that I am completely ignorant of anything to do with servers or VOIP. How hard can it be? :lol:
Flash if you go to this link https://www.ekiga.net/?page=register and register a sip account, say flash@ekiga.net and set up the account in Psip I will be able to call you. The problem with Psip at the moment it is not indicating when users are logged in. This worked perfectly fine with sipphone, another sip server. I'm not sure why this doesn't work at the ekiga sip server.

It is easy enough to connect through the sip server but it would be nice to connect without having to use a server, less dependencies.

Psip works perfectly fine as it is at the moment. We don't need to be connect to the same sip server either.

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#23 Post by smokey01 »

As I said earlier a number of the links are broken.

The help manual I created also seems to have disappeared.

I will attach a PDF version of help that should assist.

This is the link to the original thread when we first developed Psip.
http://www.murga-linux.com/puppy/viewto ... 572#212572
Attachments
Psip-Help.tar.gz
Psip Help Manual
(137.95 KiB) Downloaded 977 times

gcmartin

#24 Post by gcmartin »

Here's some helpful pics which might help one visualize SIP

Little while ago, IBM made this to easily understand. Modeling SIP via architecture.

The Architecture includes a Gateway. Your non-server product must build all of that functionality into your product versus using a server to connect.

I think what you are trying to do is to have the freedom of NOT using someone else's server for SIP registration and SIP G/W services. Is that right?
But, this is going to require some very creative thought, designing a model, and building the model+client to match your objective.

Other options are, maybe, just build your own personal local PUPPY SIP Server_Gateway, or use OpenSER or Asterisk to do all the dirty work, easily. Locally,.Asterisk will run on a PUP. Others SIP Gateway_servers can be deployed in Puppy, too.

Or, here's a thought. Suppose the Puppy Community has set up its own SIP Server_Gateway somewhere on the internet; say "http://puppy-linux.org/SIPserver" to provide call connection services needs. And suppose, that each PUP which run a Puppy SIP client registers there. And, suppose, that each PUP SIP which registers uses some "new" Puppy client so that Server_Gateway processing was shared, this would created the world's largest telephony switch. It would become enormous for Puppy and would draw every person wanting to have voice-video connections into the Puppy Community environment ...directly or indirectly. Imagine how beneficial this could be.

Hope this helps
Last edited by gcmartin on Mon 22 Aug 2011, 06:13, edited 1 time in total.

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#25 Post by Flash »

gcmartin wrote:... just build your own personal local PUPPY SIP Server_Gateway, or use OpenSER or Asterisk to do all the dirty work, easily. Locally,.Asterisk will run on a PUP. Others SIP Gateway_servers can be deployed in Puppy, too.
In my ignorant way, I was attempting to suggest something like that, only the server would run on the same machine as the VOIP client. Perhaps the Hiawatha server which comes with most Puppys could be configured to satisfy whatever the router or NAT server requires, then a SIP client would run as a separate process (if that's the right terminology) that uses the Hiawatha server. Is it even possible to run a server and a client on the same machine? Seems like it ought to be.
Or, here's a thought. Suppose the Puppy Community has set up its own SIP Server_Gateway. And suppose, that each PUP which run a Puppy SIP client registers there. And, suppose, that each PUP SIP which registers uses some "new" Puppy client so that Server_Gateway processing was shared, this would created the world's largest telephony switch. It would become enormous for Puppy and would draw every person wanting to have voice-video connections into the Puppy Community environment ...directly or indirectly. Imagine how beneficial this could be.

Hope this helps
I can see millions of people demanding help from the forum to get their VOIP working. Someone would have to come up with a really good GUI or wizard for configuring VOIP in Puppy. :lol:

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#26 Post by smokey01 »

I have spend the best part of the day updating Psip.

I have also done quite a bit of research to find a good SIP server that gives away free SIP accounts. My final decision was http://www.iptel.org

Psip seems to work very well with this server.

I have also updated the Psip help from the help menu. Be sure to have a read it has some useful information in it.

I have attached a single file called psip_gui which need to be copied to /usr/local/psip. I have zipped it up to make it upload compliant.

WARNING do back up your original psip_gui first. Just because it works here doesn't mean it's guaranteed to work for you.
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psip-0.2.tar.gz
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#27 Post by Caneri »

Hi Grant,
I used your new psip-gui and all seems ok.

Psip doesn't seem to show anyone online but it may be that there is nobody online...dunno.

The old sipphone.com doesn't seem to work any longer.

Best,
Eric

EDIT: what is the latest version of psip?
I have PSIP 0.20
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gcmartin

#28 Post by gcmartin »

Ah yes, IPTEL.
Edited: Note: All IP adapters or telephones must support STUN. (2009)

Did you notice their mention of the SIP Router Project? (latest project where the forks have agreed to combine their efforts)

If would be nice if they made it very, very easy to build a local SIP server for "calling" around on your LAN. To do it ourselves, we need
  1. dedicated PC with the following
    • 256-1GB RAM
    • 2 LAN NICs
    • adapter in the PC to route calls to Plain old telephones in your homes
  2. OS (distro)
    • Small PUP with DNSMASQ instead of DHCPD
    • SIP Router Project for local call management and connection services for SIP registration for calls outside of the home and to manage SIP calls inside the home
  3. Pictorial Instruction Manual
The above is a simplistic view of a system for running in the home or small location allow the use of internal home (line1+line2 = rj11 wiring OR line1-4 = rj45 wiring) and also allows for use of LAN connected IP phones or SIP PC clients (both wired/WiFi)

There are several "free" SIP-telephony servers available for doing this, OTB.

Biggest problem with this is that MOST newbies would NOT dare attempt this because the management issues are just beyond their reach and much too complex.

If this thread/community should undertake this effort, we must make it as easy and attractive as jumping into a "pool of naked beautiful people".

Hope this understanding helps

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#29 Post by Flash »

gcmartin wrote:...It would be nice if they made it very, very easy to build a local SIP server for "calling" around on your LAN...
Perhaps I misunderstand, but Smokey01 seems to be saying he was able to connect computers directly, within his LAN, here, without needing a server. It seems that you're screwed if you try to connect two computers directly, with a router between them. Even so-called P2P applications seem to need the initial connection to be established by a server. Is that correct?

gcmartin

#30 Post by gcmartin »

Flash wrote: ... Perhaps I misunderstand, but Smokey01 seems to be saying he was able to connect computers directly, within his LAN, here, without needing a server. ...
Yes @Flash, you are correct, and I saw that too.

But, lets say you have several PC users on your LAN whose PC names are Jack, Dick, Jane and Jill. If you want to call either one of them from your PC called Flash, how do you do that?
  • Can I use their PC name
  • Can I use their SIP information
  • How would I use an IP address
  • What will happen on my PC should they happen to call me, Flash?
I think I may have already discussed the need for this with someone somewhere....EH.

This is one of the reason this thread's discussion about SIP peer-peer is wrestling with. In the past, this function for larger networks is done in the LAN's PBX (local SIP Registrar). But, in SIP peer-peer, all tables for how to talk must be maintained and kept consistent, constantly, somewhere, locally. That's what I am trying to piece together based on the little I know about the architecture. (and to be able to call yourself a "SIP" client, that has to be adherence to some base architecture.)

@Smokey01 you are helping us you provide guidance in PET use. Thanks as we progress.
Hope that shares what I was getting at.

gcmartin

#31 Post by gcmartin »

I want to share that the items I share in the thread on PSIP is not to be viewed as criticism.

It is NOT!. I am in favor of this effort and I am in favor of a Peer-to-Peer telephony (SIP) subsystem in Puppy.

I am sharing, thru questions, (tough ones in some cases) about the issues we are up against as we progress.

I have and do offer again to write a document with instructions for how to use standard SIP compliant devices and clients in a Puppy Peer-to-Peer environment.

We just need to come up with an implementation which matches some simple structure.

I am not a coder, therefore my skills that I offer, here, is in testing and documenting what's observed and what's useful.

So, if anything I have shared seems negative, or discouraging, please don't take it that way. I only mean to share items I think we may have to address as we progress in the peer-to-peer discussion.

All other SIP implementations, either via a soft/hard VOIP PBX or an external VOIP provider for SIP registration and IP call connections are already done for us. Even SIP gateway servicing for ringing real telephones in the world is done for us. Much is free, but also, many requires some type of implementations to purchase SIP devices, and adapters to interface with real telephony calling devices that users are accustomed to around the house or office..

In summary, I have been trying to make clear that there appears to be 2 issues.
One - which is already solved for us via SIP VOIP Registrars and Gateways
Two - Peer-to-peer which they don't make it easy for us to implement, directly, one user to another.

Please understand that by sharing this, I am trying to help.

Hope this helps.

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#32 Post by Flash »

The second issue is the only one I'm interested in solving. :)

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#33 Post by Lobster »

have registered at
http://www.iptel.org/service
as smokey recommends - this will be my default
sip:lobster@iptel.org

this is my backup sip address
sip:crustylobster@ekiga.net

Thanks to smokey for new help files
simplified menu etc

Will be ensuring mic and sip account works
You will probably be able to phone me in a day or two

Ok...I'll bite.
sip:caneri@ekiga.net

I'll prolly regret posting this in public but what the 'ell.

I'm on FatDog64....I may need to load another iso/version....any ideas?
Hi Eric phoned you but you were probably busy making pizza (not online)
Will try again . . .
You will be safe - have placed my SIP and yours here (we can always remove - don't expect much traffic)
http://puppylinux.org/wikka/Psippy

I don't see any reason why PSIP should not work on any Puppy that it is compiled and installed on - that is the aim for the updates if it is not working :)
Last edited by Lobster on Wed 24 Aug 2011, 09:05, edited 1 time in total.
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#34 Post by Lobster »

I have attached a single file called psip_gui which need to be copied to /usr/local/psip. I have zipped it up to make it upload compliant.


Nice job Grant 8)
the help menu is now useful
the credits will need updating
it uses javascript (as I am running noscript I got no credits)
- anyway that is for later

Ran the tests
- tests working
- got music
- next trying to send voice mail

Might be worth getting hold of
Evil20071@proxy01.sipphone.com
as he helped either in testing or coding the original version to some degree.
He might even have renounced the way of the Sith . . . :wink:

The peer2peer connect is a great possibility
- something to aim for 8)
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#35 Post by Caneri »

Hooray!!

I had a mishap with the psip.config but now it works with iptel.org on an old 409 install.

I need a newer iso as Fatdog beta5 doesn't work with psip so far.

caneri@iptel.org

Smokey..what's your number? (EDIT: found it and added to buddy list)

EDIT: I can't send voicemail to an inbox...howto?
EDIT!: hooray!!! sent voice message to smokey01...now to try and hit on lobster...bwahahahaah.
EDIT:2 this is a handy page http://www.iptel.org/service
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#36 Post by Lobster »

I added the following SIP addresses
but they went (disappeared, AWOL, gone, vanished) and no refreshing of the refresh buddies list made any difference . . . :?

Grant
sip:smokey01@ekiga.net
Lobster
sip:crustylobster@ekiga.net
sip:lobster@iptel.org
Eric (caneri forum name) running Fatdog64
sip:caneri@ekiga.net

This file or whatever is the config file is not activating an editor or
some way of manually altering settings (I am using Slackobeta1)
/usr/local/psip/pjsua.cfg.default

I stared at the code for a while, like a slice of lemon watching fish
but nothing seemed obvious . . .

We need more coding help . . . 8)
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#37 Post by smokey01 »

lobster, you will have an error in the config file.

all commands must start with --

If you make an error or a typo your buddy list will disappear and psip will not work properly.

Have a look at my help file from within Psip.

The config file is located in /root/.psip/pjsua.cfg

If you send me your config file I will take a look, correct it and send it back.

BTW you need to activate your voice mail IPTel.org then I will be able to leave you messages.

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#38 Post by Caneri »

Here is a working psip.config

Code: Select all

#
# Network settings:
#
--local-port 5060
#I think ice is important for maintaining routing information for those with dynamic internet ip addresses
--use-ice
#I think this might only work if you have a gizmo account
#--stun-srv=stun01.sipphone.com

#
# Media settings:
#
# using default --clock-rate 12000
--quality 10
# using default --ec-tail 200
# using default --ilbc-mode 20
--rtp-port 4000

#
# User agent:
#
--max-calls 4

#
# Buddies:
--add-buddy sip:crustylobster@ekiga.net
--add-buddy sip:520@ekiga.net
--add-buddy sip:lobster@iptel.org
--add-buddy sip:smokey01@iptel.org
--add-buddy sip:1001@iptel.org

#












# Account 0:
--id sip:usrid@iptel.org
--registrar sip:iptel.org
--realm *
--username (your userid from whatever sip provider you sign up with)
--password xxxxxxxxxxxx
--reg-timeout 55
Always add one blank line at the end of the psip.cfg.
The cfg will not work without the extra blank line at the end.

I use call quality 10...this seems to improve the playback.

@smokey...got the voice messages and now have my usb headst working...should be better sound quality.
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OTB Configuration Generator

#39 Post by gcmartin »

Would it be helpful if we had an OTB Configuration Generator for PSIP? A tool like this would also be a syntax checker as well.

If so, who could design such?
Should it be integrated into PSIP OR should it be a companion to PSIP?

I think this would make it easier for us (and newbies) to use without having to appeal for assistance.

Hope this helps

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Re: OTB Configuration Generator

#40 Post by smokey01 »

gcmartin wrote:Would it be helpful if we had an OTB Configuration Generator for PSIP? A tool like this would also be a syntax checker as well.

If so, who could design such?
Should it be integrated into PSIP OR should it be a companion to PSIP?

I think this would make it easier for us (and newbies) to use without having to appeal for assistance.

Hope this helps
Psip already has one, of sorts. The first time Psip is run it does not have a pjsua.cfg file in /root/.psip. It copies a default one from /usr/local/psip which is called pjsua.cfg.default and renames it to pjsua.cfg.

It's a little hard to automate it more it than this as you need to add your details, servers and buddy's.

This can all be done manually with a text editor but it can also be done under menu item configure.

The error checking is another issue but I'm not sure who has the skills to address this.

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